Delay reduction method for telephony systems with multiple packet generators

ABSTRACT

A telephony system and method is provided that reduces delay and provides better utilization of upstream bandwidth in delivering packet telephony services to a plurality of subscriber lines via a cable modem system. An exemplary system includes a plurality of voice processing modules, a host processor, and a buffer. Each voice processing module receives digital voice signals from a separate set of subscriber lines, compresses the digital voice signals to generate a voice packet, and transfers the voice packet to the buffer. The host processor then assembles a packet by concatenating the voice packets and transmits the assembled packet for delivery over a data network. Because the plurality of voice processing modules process the voice packets in parallel, delay is reduced in the assembly and transmission of the assembled packet.

CROSS-REFERENCE TO RELATED APPLICATIONS

[0001] This application claims priority to the following provisionalapplication:

[0002] U.S. Patent Ser. No. 60/209,551, entitled “Delay Reduction Methodfor Telephony Systems with Multiple Packet Generators,” filed Jun. 6,2000, by Rabenko, (still pending) (incorporated by reference in itsentirety herein).

BACKGROUND OF THE INVENTION

[0003] 1. Field of the Invention

[0004] The present invention is directed to telephony systems. Moreparticularly, the present invention is directed to systems and methodsfor providing telephony services to a plurality of subscriber lines overa packet network.

[0005] 2. Background

[0006] High speed data networks, such as the Internet, have emerged asviable platforms for the delivery of telephony services. For example,cable operators are currently utilizing hybrid fiber-coaxial (HFC)networks to deliver packet telephony services to subscribers viaresidential cable modems. It is anticipated that cable modem systemswill enable the deployment of telephony services in a manner that isless costly than existing circuit-switched alternatives, as well aspermit the delivery of unique value-added features, such as integratedvoice mail and e-mail messaging.

[0007] The North American telephone market includes both single familydwellings and multiple dwelling units. According to conventionalindustry usage, the abbreviation “MDU” is used to refer both to multipledwelling units as well as to the telephony equipment used to servicethem. For the sake of clarity, throughout the rest of this document theterm “MDU” will be used exclusively to refer to multiple dwelling unitsthemselves, whereas the terms “MDU equipment,” “MDU system,” or “MDUtelephony system” will be used to refer to the telephony equipment usedin servicing them.

[0008] The demand for subscriber circuits in MDU applications comprisesapproximately 30% of all telephony installations. However, conventionaldeployments of voice telephony services using cable modem systems aresupported entirely by equipment designed to meet the requirements ofsingle family dwellings. For example, a conventional cable modem deviceadapted for delivering voice telephony services over an HFC network isdesigned to support only a limited number of subscriber lines, such asfour subscriber lines, per cable modem. (As used herein, the term“subscriber line” generally refers to the line that connects acustomer's telephone to one or more interfaces to a high speed datanetwork.) Consequently, a device of this type is not suitable fordeploying telephony services in an MDU with a large number of subscriberlines, such as in an apartment or other high-density structure. Althoughmultiple devices of this type could be used to support a greater numberof subscriber lines, such a deployment would be prohibitively expensivein light of the cost of the devices.

[0009] Furthermore, conventional cable modem devices for deliveringvoice telephony services over an HFC network utilize a single digitalsignal processor (DSP) for processing voice signals from one or moresubscriber lines for transmission over the HFC network. Because a singleDSP is used, each voice channel must be processed serially. Thus, ifconventional devices were utilized to support a larger number ofsubscriber lines, the single DSP would act as a bottleneck, causingtransmission delay that would cause a reduction in Quality of Service(QOS). A much more powerful DSP would have to be utilized in order toprocess a larger number of subscriber lines while maintaining atransmission rate that would not cause call quality to suffer. The useof a very powerful DSP, however, would cause a dramatic increase in thecost overall cost of the device.

[0010] What is desired, then, is a system and method for deliveringpacket telephony services via a cable modem system that is capable ofsupporting a greater number of subscriber lines per cable modem.Furthermore, the desired system and method should reduce delay in theprocessing of voice packets and provide for improved bandwidthutilization so that a satisfactory packet transmission rate, andtherefore QOS, may be maintained. In addition, the desired system andmethod should be cost efficient, providing more attractive cost per linecharacteristics than conventional systems and methods for deliveringpacket telephony services via a cable modem system.

BRIEF SUMMARY OF THE INVENTION

[0011] The present invention is directed to a telephony system andmethod that reduces delay and improves network bandwidth utilization indelivering packet telephony services to a plurality of subscriber lines.Embodiments of the present invention permit such services to bedelivered in a cost-efficient manner.

[0012] In embodiments, the system includes a first audio processingmodule and a first voice processing module coupled to a first set ofsubscriber lines, a second audio processing module and a second voiceprocessing module coupled to a second set of subscriber lines, a hostprocessor, and a buffer. The first audio processing module receivesfirst analog voice signals from one of the first set of subscriber linesand converts the first analog voice signals into first digital voicesignals. The first voice processing module receives the first digitalvoice signals, compresses them to generate a first voice packet, andtransfers the first voice packet to the buffer. The second audioprocessing module receives second analog voice signals from one of thesecond plurality of subscriber lines and converts the second analogvoice signals into second digital voice signals. The second voiceprocessing module receives the second digital voice signals, compressesthem to generate a second voice packet, and transfers the second voicepacket to the buffer.

[0013] The host processor then assembles a packet from the first voicepacket and the second voice packet and transmits the assembled packetfor delivery over a data network. In embodiments of the presentinvention, the first voice processing module and the second voiceprocessing module generate the first voice packet and the second voicepacket in parallel, thereby reducing delay in the assembly andtransmission of the assembled packet.

[0014] In further embodiments of the present invention, the hostprocessor transmits the assembled packet for delivery over an HFCnetwork. Furthermore, the host processor may transmit the assembledpacket for delivery over an HFC network during an assigned upstreamburst opportunity.

[0015] In alternate embodiments, the first voice processing module andthe second voice processing module each comprise a digital signalprocessor.

[0016] The invention is advantageous in that it permits packet telephonyservices to be delivered to a plurality of subscriber lines via a singlecable modem.

[0017] The invention is further advantageous in that it reduces delay inthe delivery of packet telephony services to a plurality of subscriberlines via a cable modem system.

[0018] The invention is also advantageous in that it provides forimproved utilization of network bandwidth in delivering packet telephonyservices to a plurality of subscriber lines via a cable modem system.

[0019] Another benefit of the invention is that it provides areduced-cost alternative for providing packet telephony services to aplurality of subscriber lines via a cable modem system.

[0020] Additional features and advantages of the invention will be setforth in the description that follows, and in part will be apparent fromthe description, or may be learned by practice of the invention. Theobjectives and other advantages of the invention will be realized andattained by the system and method particularly pointed out in thewritten description and claims hereof as well as the appended drawings.

BRIEF DESCRIPTION OF THE DRAWINGS/FIGURES

[0021] The accompanying drawings, which are incorporated herein and forma part of the specification, illustrate the present invention and,together with the description, further serve to explain the principlesof the invention and to enable a person skilled in the pertinent art tomake and use the invention.

[0022]FIG. 1 depicts an exemplary telephony system within whichembodiments of the present invention may operate.

[0023]FIGS. 2A, 2B and 2C illustrate alternate example MDU telephonysystems in accordance with embodiments of the present invention.

[0024]FIG. 3 illustrates upstream burst timing for a cable modem portionof an MDU telephony system in accordance with embodiments of the presentinvention.

[0025]FIG. 4 illustrates an assembled packet for upstream transmissionin accordance with embodiments of the present invention.

[0026]FIG. 5 depicts a portion of an example MDU telephony system inaccordance with embodiments of the present invention.

[0027]FIG. 6 is a flowchart showing a method for delay reduction in atelephony system in accordance with embodiments of the presentinvention.

[0028] The present invention will now be described with reference to theaccompanying drawings. In the drawings, like reference numbers indicateidentical or functionally similar elements. Additionally, the left-mostdigit(s) of a reference number identifies the drawing in which thereference number first appears.

DETAILED DESCRIPTION OF THE INVENTION

[0029] A. Overview of the Invention

[0030] The present invention is generally directed to a system andmethod for delay reduction and improved bandwidth utilization in thedelivery oftelephony services over packet networks to a plurality ofsubscriber lines. The present invention is particularly applicable topacket networks with reservation-based transmission capabilities, suchas cable modem systems and fixed wireless systems, as will be describedin detail herein. For example, the present invention will reduce delayand improve bandwidth utilization in a cable modem system wherein asingle cable modem is utilized to deliver voice telephony services to aplurality of subscriber lines. Accordingly, the present invention may beadvantageously utilized to deliver telephony services to multipledwelling units (MDUs).

[0031] B. Example Operating Environment

[0032]FIG. 1 depicts an exemplary telephony system 100 within whichembodiments of the present invention may operate. The exemplarytelephony system 100 permits telephone communication to be carried outbetween any of a first plurality of telephony devices 102 a through 102n, depicted on the left side of FIG. 1, and any of a second plurality oftelephony devices 118 a through 1118 n, depicted on the right side ofFIG. 1. The telephony devices 102 a through 102 n and 118 a through 118n may include telephones, facsimile machines, or any other type oftelephony device adapted for communicating voice signals over acircuit-switched or packet-switched network. For the purposes of thisexample, each of the telephony devices 118 a through 118 n are assumedto be attached to a different subscriber line within the same MDU.

[0033] As shown in FIG. 1, the exemplary telephony system 100 includesboth a public switched telephone network (PSTN) 104 for providingcircuit-switched telephony services to telephony devices 102 a through102 n and a packet telephony system 120 for providing packet-basedtelephony services to telephony devices 118 a through 118 n at the MDU.The PSTN 104 and the packet telephony system 120 are connected by meansof a network gateway 106. The network gateway 106 facilitates telephonecommunication between the networks by operating to convert analog voicesignals received from the PSTN 104 to digital voice packets suitable fortransmission via the packet telephony system 120 and to convert digitalvoice packets received from the packet telephony system 120 to analogvoice signals suitable for transmission via the PSTN 104.

[0034] The packet telephony system 120 provides for the communication ofvoice packets based on a bi-directional transfer of Internet protocol(IP) traffic between a packet network 108, which may include theInternet, and an MDU telephony system 114, which delivers voicetelephony services to the plurality of telephony devices 118 a-118 n. Tothis end, the packet telephony system 120 incorporates a cable modemsystem comprising a cable modem termination system (CMTS) 110, a hybridfiber-coaxial (HFC) network 112, and a cable modem 116. As depicted inFIG. 1, the cable modem 116 is an integral part of the larger MDUtelephony system 114 that controls the processing and transmission ofvoice packets to and to and from the plurality of subscriber linessupporting telephony devices 118 a through 118 n.

[0035] The CMTS 110 is a device typically located at a cable headendthat controls the upstream and downstream transfer of data betweenitself and the cable modem 116, as well as any other cable modems towhich it may be attached by means of the HFC network 112. (In thecontext of a cable modem system, the term “downstream” generally refersto a transmission from the CMTS to a cable modem, while the term“upstream” generally refers to a transmission from a cable modem to theCMTS). In particular, the CMTS 110 controls the upstream transfer ofinformation by assigning short periodically-scheduled transmissionopportunities to the cable modem 116. Because the cable modem 116 canonly transmit information during these reserved time periods, cablemodem systems may be considered reservation-based transmission systems.The CMTS 110 further operates to modulate and terminate RF signals goingto and coming from the HFC network 112, and bridges these to a moregeneric type of data transport to connect with the packet network 108.

[0036] The HFC network 112 provides for the high-speed, reliable, andsecure transport of data between the CMTS 110 at the cable headend andthe cable modem 116 at the MDU. As will be understood by persons ofordinary skill in the relevant art(s), the HFC network 112 may comprisecoaxial cable, fiberoptic cable, or a combination of coaxial cable andfiberoptic cable linked via one or more fiber nodes.

[0037] The cable modem 116 is a device within the MDU telephony system114 that operates as an interface between the plurality of customersubscriber lines attached to the telephony devices 118 a through 1118 nand the HFC network 112 for the delivery of packet telephony services.In particular, the cable modem 16 transfers voice packets to and fromthe HFC network 112 in compliance with the DOCSIS specificationpublished by CableLabs.

[0038] As shown in the exemplary telephony system 100, the single cablemodem 116 may be required to support a potentially large number ofsubscriber lines. Accordingly, the MDU telephony system 114 of thepresent invention is designed to reduce delay in the processing of voicepackets from the plurality of subscriber lines and to provide forimproved upstream bandwidth utilization in transferring voice packetsfrom the MDU telephony system 114 to the CMTS 110 over the HFC network112. Embodiments of the present invention thereby ensure acceptabletelephone call quality by maintaining a sufficiently high packettransmission rate. Furthermore, embodiments of the present invention mayachieve these goals in a cost efficient manner. An MDU telephony systemin accordance with the present invention will now be described in moredetail.

[0039] C. MDU Telephony System in Accordance with Embodiments of thePresent Invention

[0040] An MDU telephony system in accordance with embodiments of thepresent invention will reduce delay and improve upstream bandwidthutilization where telephony services are delivered to multiplesubscriber lines via a single cable modem. As will be described herein,these objectives are achieved, in part, through the use of multipleaudio processing modules and voice processing modules for simultaneouslygenerating voice packets from a plurality of subscriber lines, inconjunction with the use of concatenation techniques for combiningmultiple voice packets into a single upstream burst for transmission.Accordingly, embodiments of the present invention are well-suited forproviding packet telephony services to a MDU.

[0041]FIG. 2A illustrates an exemplary MDU telephony system 200 inaccordance with embodiments of the present invention. The exemplary MDUtelephony system 200 operates to efficiently deliver packet telephonyservices to up to 8 subscribers simultaneously. However, as will bediscussed in more detail herein, the present invention is capable ofsupporting many more subscriber lines. The size of the MDU telephonysystem 200 is limited in this description for the sake of clarity.

[0042] The example MDU telephony system 200 includes a plurality oftelephone connection interfaces 202 a-202 h, a plurality of subscriberline interface circuits (SLICs) 204 a-204 h, a voice and audioprocessing module (VAPM) 206, an integrated VAPM and cable modem (CM)208, and a cable tuner 210.

[0043] The telephone connection interfaces 202 a-202 h each comprise adevice for connecting a telephone, or other telephony device, to the MDUtelephony system 200. In embodiments, each telephone connectioninterface 202 a-202 h comprises a standard RJ-11 jack for connecting aPOTS (Plain Old Telephone Service) telephony device, such as a POTSphone or fax machine, to a subscriber line for the delivery of telephonyservices. However, as will be appreciated by persons skilled in the art,various other devices may be used to implement the telephone connectioninterfaces 202 a-202 h, including but not limited to any appropriateconnection means for connecting a telephony device to a subscriber line.

[0044] As illustrated in FIG. 2A, each of the telephone connectioninterfaces 202 a-202 h is coupled via a subscriber line to acorresponding SLIC 204 a-204 h. In embodiments, each SLIC 204 a-204 ecomprises a well-known integrated circuit for performing some or all ofthe POTS interface functions used in delivering standard POTS service toa telephony device attached to each of the telephone connectioninterfaces 202 a-202 h. For example, in embodiments, each SLIC 202 a-202h provides power and ringing signals to the attached subscriber line,provides signaling functionality, and monitors subscriber lineconditions.

[0045] As further illustrated in FIG. 2A, the four SLICs 204 a, 204 b,204 c, and 204 d are each coupled to the integrated VAPM and CM 208 andthe four remaining SLICs 204 e, 204 f, 204 g and 204 h are each coupledto the VAPM 206. Both the integrated VAPM and CM 208 and the VAPM 206comprise an audio processing module and a voice processing module thatoperate to convert analog voice signals received from the SLICs intodigital voice packets for delivery over an HFC network (not shown) and,conversely, to convert digital voice packets received from the HFCnetwork to analog voice signals for transmission over the plurality ofsubscriber lines.

[0046] In particular, the integrated VAPM and CM 208 comprises an audioprocessing module 216, a voice processing module 218, and a cable modemmodule 220. The audio processing module 216 performs theanalog-to-digital (A/D) conversion of voice signals received from theSLICs 204 a, 204 b, 204 c, and 204 d, and the digital-to-analog (D/A)conversion of voice signals received from the voice processing module218. In embodiments, the audio processing module 216 comprises fourCODECs, each of which corresponds to one of the four voice channelsassociated with the SLICs 204 a, 204 b, 204 c and 204 d., for performingD/A and A/D conversion of voice signals. In further embodiments, eachCODEC comprises an integrated circuit for performing signal conversionfunctions.

[0047] The voice processing module 218 performs the compression ofdigital voice signals received from the audio processing module 216 andthe decompression of digital voice signals received from the cable modemmodule 220. In embodiments, the voice processing module 218 comprises adigital signal processor for performing compression and decompression ofdigital voice signals. In further embodiments, the voice processingmodule 218 performs compression and decompression in accordance with oneor more standard compression/decompression techniques, including but notlimited to the G.711, G.723.1, G.726 and G.729 standards.

[0048] The cable modem module 220 within the integrated VAPM and CM 208includes a cable modem and additional components for providing aninterface to the HFC network and for bridging the voice packets to thedata network for transfer in compliance with the DOCSIS specification.In particular, the cable modem module 220 operates to transfer voicepackets from the voice processing module 218 to the HFC network and fromthe HFC network to the voice processing module 218. As shown in FIG. 2A,the cable modem module 220 transfers voice packets to and from theexternal HFC network via the cable tuner 210. In embodiments, the cabletuner 210 is a CMOS tuner.

[0049] In embodiments, the integrated VAPM and CM 208 comprises aBroadcom BCM 3352 QAMLINK™ Single-Chip 4-Channel VoIP ResidentialGateway, manufactured by Broadcom Corporation of Irvine, Calif.

[0050] Like the integrated VAPM and CM 208, the VAPM 206 also includesan audio processing module (audio processing module 212) and a voiceprocessing module (voice processing module 214) for processing voicepackets for delivery to and from any of four different subscriber lines.The audio processing module 212 and the voice processing module 214 aresubstantially the same as the audio processing module 216 and the voiceprocessing module 218 described in reference to the integrated VAPM andCM 208, above, except that the audio processing module 212 and the voiceprocessing module 214 process voice data for transfer between the SLICs204 e, 204 f, 204 g and 204 h and the cable modem module 220, asillustrated in FIG. 2A. Packets are communicated between the VAPM 206and the integrated VAPM and CM 208 via an external expansion bus 222.

[0051] In embodiments, the VAPM 206 comprises a Broadcom MDU DSP, partnumber BCM 3341, manufactured by Broadcom Corporation of Irvine, Calif.

[0052] It should be noted that the present invention is not limited toembodiments in which the audio processing module 216 and the voiceprocessing module 218 are incorporated with a cable modem as part of asingle integrated device. For example, FIG. 2B illustrates an alternateembodiment of the MDU telephony system 200 in which the audio processingmodule 216 and the voice processing module 218 may comprise a separateVAPM 224 that is coupled to a cable modem device 208′ via the expansionbus 222. In the alternate embodiment of the MDU telephony system 200 ofFIG. 2B, the cable modem device 208′ may comprise a Broadcom BCM 3350QAMLINK™ Single Chip Cable Modem or any of the Broadcom integrated cablemodem devices designated with part numbers BCM 3345, BCM 3360 or BCM3370, manufactured by Broadcom Corporation of Irvine, Calif.Additionally, in the alternate embodiment of the MDU telephony system200 of FIG. 2B, the VAPM 224 may comprise a Broadcom BCM MDU DSP, partnumber BCM 3341, manufactured by Broadcom Corporation of Irvine, Calif.

[0053] As will be described further herein, the use of multiple audioprocessing modules, such as the audio processing modules 212 and 216,and multiple voice processing modules, such as the voice processingmodules 214 and 218, in the example MDU telephony system 200 permitvoice packets to be generated in parallel from the first set ofsubscriber lines 202 a-d and from the second set of subscriber lines 202e-202 h. These voice packets are then concatenated into a singleassembled packet for upstream transmission by the cable modem module220.

[0054]FIG. 2C illustrates an alternate exemplary MDU telephony system230 in accordance with embodiments of the present invention. Thealternate exemplary MDU telephony system 230 is similar to the system200 illustrated in FIG. 2A, except that the exemplary MDU telephonysystem 230 has been adapted to deliver packet telephony services to upto 12 subscribers simultaneously, rather than 8. This has been achievedthrough the use of two VAPMs, 206 a and 206 b, rather than one. Each ofthe VAPMs, 206 a and 206 b, comprises a separate audio processing moduleand a voice processing module for processing four channels of voice dataand is essentially the same as the VAPM 206 described in reference toFIG. 2A, above. Both VAPMs 206 a and 206 b are coupled to the cablemodem module 220 via the expansion bus 222.

[0055] As illustrated in FIG. 2C, the use of an additional VAPM allowsfor four more subscriber lines to be supported by the exemplary MDUtelephony system 230. These four additional subscriber lines areimplemented through the use of four additional telephone connectioninterfaces 202 i, 202 j, 202 k, and 202 l, and four additional SLICs 204i, 204 j, 204 k and 204 i. However, the present invention is not limitedto a particular number of subscriber lines, a particular number ofVAPMs, or a particular ratio of subscriber lines to VAPMs. As will beappreciated by persons of ordinary skill in the art from the teachingsprovided herein any number of subscriber lines, VAPMs, and subscriberlines per VAPM can be used to practice the present invention.

[0056] D. Voice Packet Processing and Transmission in Accordance withEmbodiments of the Present Invention

[0057] The technique by which voice packets are processed andtransmitted in accordance with embodiments of the present invention willnow be described. The description will be made with continued referenceto the exemplary MDU telephony system 230 of FIG. 2C. As discussedabove, the exemplary MDU telephony system 230 efficiently deliverspacket telephony services to up to 12 subscriber lines through the useof an integrated VAPM and CM 208 and two VAPMs 206 a and 206 b. However,the technique of the present invention is not limited to this exemplaryembodiment.

[0058] In accordance with the DOCSIS specification, a cable modem mayonly transmit data upstream in short transmissions, called bursts,during transmission opportunities assigned to it by the CMTS. Wheretelephony services are being provided, it is anticipated that the burstopportunities will arrive at a dependable periodic interval. FIG. 3 is atiming diagram 300 that illustrates an example series of upstream burstopportunities for the cable modem module 220 within the integrated VAPMand CM 208. The timing diagram 300 shows three example burstopportunities, 302, 304 and 306, occurring at times G_(i), G_(i+1) andG_(i+2), which together define a dependable periodic interval. Toutilize each burst opportunity the cable modem module 220 must have dataready for transmission before times G_(i), G_(i+1) and G_(i+2). Thus,the finite intervals between the burst opportunities 302, 304 and 306delineate the rate at which data may be transmitted upstream by thecable modem module 220. In embodiments of the present invention, thetransmission intervals may be roughly 10 milliseconds apart.

[0059] Embodiments of the present invention take advantage of the factthat DOCSIS bursts arrive at a dependable periodic interval to align theprocessing and assembly of voice packets from multiple voice processingmodules with the scheduled transmission opportunities. By concatenatingthe voice packets into a single DOCSIS packet, an MDU telephony systemin accordance with the present invention is capable of transmittingmultiple telephone channels during a single DOCSIS upstream burstinstead of one, thereby providing for better utilization of upstreambandwidth.

[0060] For example, the MDU telephony system 230 of FIG. 2C is capableof transmitting three telephone channels during a single DOCSIS upstreamburst. This is because the MDU telephony system 230 is capable ofsimultaneously processing and assembling a voice packet from each of thethree different voice processing modules associated with the integratedVAPM and CM 208, the VAPM 206 a and the VAPM 206 b. In particular, theMDU telephony system 230 periodically assembles a packet for upstreamtransmission that includes a plurality of voice packets: one from thevoice processing module 218 within the integrated VAPM and CM 208, onefrom the voice processing module within the first VAPM 206 a, and onefrom the voice processing module within the second VAPM 206 b.

[0061]FIG. 4 illustrates the format of an example assembled packet 400for upstream transmission in accordance with embodiments of the presentinvention. The example assembled packet 400 includes physical layer(PHY) overhead 402, media access layer (MAC) overhead 404, a first voicepacket 406, a second voice packet 408, and a third voice packet 410.

[0062] The PHY overhead 402 comprises information necessary for the RFtransmission and reception of the burst and may include, for example, apreamble or training sequence that permits a CMTS at the cable headendto “lock on” to the burst and demodulate the transmitted signal. The MACoverhead 404 comprises header information necessary for transferringdata in accordance with the DOCSIS protocol, and may include, forexample, a frame control field (1 byte), a MAC_PARM field (1 byte), aLEN (SID) field (2 bytes), an EHDR field (0-240 bytes), and a HeaderCheck Sequence (HCS) field (2 bytes). The PHY overhead 402 and the MACoverhead 404 are both defined in the DOCSIS 1.1 specification and arewell known in the art.

[0063] As shown in FIG. 4, the payload of the example assembled packet400 includes three voice packets concatenated together for transmissionin a single upstream burst: a first voice packet 406, a second voicepacket 408, and a third voice packet 410. These voice packets may beconcatenated in accordance with the concatenation provisions of theDOCSIS 1.1 specification.

[0064] In accordance with embodiments of the present invention, each ofthese voice packets originates from a different voice processing moduleservicing a different set of subscriber lines. For example, the firstvoice packet 406 may originate from the voice processing module 218within the integrated VAPM and CM 208 that services the four subscriberlines coupled to the SLICs 204 a, 204 b, 204 c, and 204 d. Likewise, thesecond voice packet 408 may originate from the voice processing modulewithin the VAPM 206 a that services the four subscriber lines coupled tothe SLICs 204 e, 204 f, 204 g and 204 h. Finally, the third voice packet410 may originate from the voice processing module within the VAPM 206 bthat services the four subscriber lines coupled to the SLICs 204 i, 204j, 204 k, and 204 l.

[0065] By concurrently processing voice packets from more than one voiceprocessing module, embodiments of the present invention permit aplurality of voice channels to be transmitted in a single upstream burstwithout requiring a concomitant increase in processing speed for any ofthe voice processing modules. For example, in regard to the MDUtelephony system 230, three voice channels may be transmitted in asingle upstream burst. However, each of the three voice processingmodules are required to contribute only as single voice packet inadvance of each burst opportunity. Accordingly, in embodiments of thepresent invention, each of the voice processing modules may beimplemented using processors sized as though the system were one thirdsmaller, resulting in a more cost-efficient design.

[0066]FIG. 5 depicts a portion of an MDU telephony system in accordancewith embodiments of the present invention. The portion of the MDUtelephony system illustrated in FIG. 5 includes a host processor 502, atransmission buffer memory 504, the voice processing module 218, whichis part of integrated VAPM and CM 208, a voice processing module 506,which represents the voice processing module within the VAPM 206 a, anda voice processing module 508, which represents the voice processingmodule within the VAPM 206 b.

[0067] The host processor 502 resides within the cable modem module 220and controls the transfer of voice packets from the three voiceprocessing modules, as well as the assembly and upstream transmission ofassembled packets over the HFC network. In alternate embodiments of thepresent invention, the host processor 502 resides within the integratedVAPM and CM 208 but is located external to the cable modem module 220.In further embodiments, the host processor 502 comprises a 32-bit MIPS®processor. However, the invention is not so limited, and the hostprocessor 502 may comprise any suitable microprocessor for performingthe functions described herein.

[0068] The transmission buffer memory 504 comprises a memory utilized bythe host processor for assembling packets for upstream transmission. Inembodiments, the transmission buffer memory 504 resides within the cablemodem module 220. In alternate embodiments, the transmission buffermemory 504 resides in an external memory module coupled to theintegrated VAPM and CM 208. In either case, the transmission buffermemory 504 is accessible to the host processor 502 as well as the voiceprocessing modules 218, 506, and 508, as shown in FIG. 5.

[0069] In accordance with the technique of the present invention, thehost processor 502 collects voice packets from the voice processingmodules 218, 506, and 508 at periodic intervals, assembles them into anassembled packet in a buffer within the transmission buffer memory 504,and transmits the assembled packet upstream during an assigned upstreamburst opportunity.

[0070] The host processor 502 assembles the packet for upstreamtransmission by constructing the necessary header information in abuffer within the transmission buffer memory 504 and controlling thetransfer of voice packets from the voice processing modules 218, 506 and508.

[0071] In embodiments, the host processor 502 controls the transfer ofvoice packets from the voice processing modules 218, 506 and 508 usingDMA (Direct Memory Access) transfers. In such embodiments, since thehost processor 502 is responsible for assembling the packet for upstreamtransmission in accordance with the DOCSIS specification, the hostprocessor calculates the overall length of the assembled packet anddetermines the necessary starting location for each voice packet thatwill comprise a portion of its payload. The host processor 502 thenassigns a DMA pointer to each voice processing module that points to thepredetermined starting location for each voice packet in the bufferwithin the transmission buffer memory 504. After each voice processingmodule has completed processing of a voice packet, it will perform a DMAtransfer of the voice packet to the transmission buffer memory 504.

[0072] In alternate embodiments, the host processor 502 controls thetransfer of voice packets from the voice processing modules 218, 506 and508 using segmented, or linked-list, DMA transfers. As will beappreciated by those of ordinary skill in the pertinent art(s), in suchembodiments, the voice processing modules transfer the voice packets tonon-contiguous areas within the transmission buffer memory 504 usingpointers provided by the host processor 502. The voice packets aresubsequently linked together by the host processor 502 to generate anassembled packet for upstream transmission.

[0073] In embodiments of the present invention, the voice processingmodules 218, 506 and 508 operate in parallel, so that each will generatea voice packet for upstream transmission during substantially the sametime period. Consequently, the combined delay attributable to the voiceprocessing modules 218,506, and 508 will be only the time required for asingle voice processing module to generate a voice packet.

[0074] Once a complete packet comprising a header and a payloadincluding three voice packets (one from each voice processing module)has been assembled in the transmission buffer memory 504, the hostprocessor 502 then causes the assembled packet to be transferred fromthe transmission buffer memory 504 to the necessary upstreamtransmission equipment for transmission over the HFC network. Inembodiments, the assembly of the voice packet is temporally aligned withthe arrival of an assigned upstream burst opportunity, such that theassembled packet is transmitted during the assigned burst opportunitythat arrives after assembly has completed.

[0075]FIG. 6 depicts a flowchart 600 of a method for delay reduction inan MDU telephony system in accordance with embodiments of the presentinvention. The invention, however, is not limited to the descriptionprovided by the flowchart 600. Rather, it will be apparent to personsskilled in the art from the teachings provided herein that otherfunctional flows are within the scope and spirit of the presentinvention. The flowchart 600 will be now described with continuedreference to the example MDU telephony system 230 depicted in FIG. 2Cand FIG. 5.

[0076] At step 602, a first set of analog voice signals is received fromone of a first plurality of subscriber lines. In the example MDUtelephony system 230, this step is performed, for example, when theaudio processing module 216 within the integrated VAPM and CM 208receives analog voice signals from one of four subscriber lines via theSLICs 202 a, 202 b, 202 c or 202 d.

[0077] At step 604, a second set of analog voice signals is receivedfrom one of a second plurality of subscriber lines. In the example MDUtelephony system 230, this step is performed, for example, when theaudio processing module within the VAPM 206 a receives analog voicesignals from one of four subscriber lines via the SLICs 202 e, 202 f,202 g or 202 h.

[0078] At step 606, the first set of analog voice signals are convertedinto first digital voice signals. In the example MDU telephony system230, this step is performed, for example, when the audio processingmodule 216 within the integrated VAPM and CM 208 converts the analogvoice signals received from one of four subscriber lines via the SLICs202 a, 202 b, 202 c or 202 d into digital voice signals for transfer tothe voice processing module 218.

[0079] At step 608, the second set of analog voice signals are convertedinto second digital voice signals. In the example MDU telephony system230, this step is performed, for example, when the audio processingmodule within the VAPM 206 a converts the analog voice signals receivedfrom one of four subscriber lines via the SLICs 202 e, 202 f, 202 g and202 h into digital voice signals for transfer to the voice processingmodule 506 within the VAPM 206 a.

[0080] At step 610, the first digital voice signals are compressed intoa first voice packet. In the example MDU telephony system 230, this stepis performed, for example, when the voice processing module 218 withinthe integrated VAPM and CM 208 compresses the digital voice signalsreceived from the audio processing module 216 into a voice packet.

[0081] At step 612, the second digital voice signals are compressed intoa second voice packet. In the example MDU telephony system 230, thisstep is performed, for example, when the voice processing module 506within the VAPM 206 a compresses the digital voice signals received fromthe audio processing module within the VAPM 206 a into a voice packet.

[0082] At step 614, the first voice packet is transferred into a buffer.In the example MDU telephony system 230, this step is performed, forexample, when the voice processing module 218 within the integrated VAPMand CM 208 transfers the generated voice packet to the transmissionbuffer memory 504 also within the integrated VAPM and CM 208. Inembodiments, this transfer is a DMA transfer. Where DMA transfers areused, the host processor 502 provides a pointer to the voice processingmodule 218 in advance of this step that indicates the appropriatedestination address in the transmission buffer memory 504.

[0083] At step 616, the second voice packet is transferred into thebuffer. In the example MDU telephony system 230, this step is performedfor example, when the voice processing module 506 within the VAPM 206 atransfers the generated voice packet to the transmission buffer memory504 within the integrated VAPM and CM 208. In embodiments, this transferis a DMA transfer. Where DMA transfers are used, the host processor 502provides a pointer to the voice processing module 506 in advance of thisstep that indicates the appropriate destination address in thetransmission buffer memory 504.

[0084] As shown in FIG. 6, the method steps 602, 606, 610 and 614 may beperformed in parallel with the method steps 604, 608, 612 and 616. Asdiscussed in particular above, in embodiments of the present invention,the compression steps 610 and 612 are performed roughly simultaneouslyin order to reduce delay in the assembly and transmission of anassembled packet that will ultimately contain a voice packet from eachvoice processing module.

[0085] At step 618, a packet is assembled for upstream transmission fromthe first voice packet and the second voice packet in the buffer. In theexample MDU telephony system 230, this step is performed, for example,when the host processor 502 assembles a packet in the transmissionbuffer memory 504 that includes a voice packet transferred from thevoice processing module 218 and a voice packet transferred from thevoice processing module 506. As discussed above, in the example MDUtelephony system 230, a third voice packet may also included in theassembled packet from a third voice processing module. In accordancewith embodiments of the present invention, any number of voice packetsmay be assembled for transmission using the techniques described herein,within the limits of the transmission system. For example, the limit onpacket size imposed by TCP/IP is 1500 bytes.

[0086] At step 620, the assembled packet is transmitted for deliveryover a data network. In the example MDU telephony system 230, this stepis performed, for example, when the host processor 502 causes theassembled packet to be transferred from the transmission buffer memory504 to the necessary upstream transmission equipment for transmissionover the HFC network. In embodiments, the assembly of the voice packetis temporally aligned with the arrival of an assigned upstream burstopportunity, such that the assembled packet is transmitted during theassigned upstream burst opportunity that arrives after assembly hascompleted.

[0087] In accordance with the above-described system and method, asingle DOCSIS upstream burst is used to transmit a plurality oftelephone channels instead of one. Consequently, embodiments of thepresent invention will reduce the PHY and MAC level overhead per voicepacket transfer and permit better upstream channel utilization. As willbe appreciated by persons of ordinary skill in the art, although anembodiment of the present invention has been described in which threetelephone channels are transmitted in a single burst, the invention isnot so limited, and any number of telephone channels could betransmitted within the limitations of the transmission system.

[0088] Embodiments of the present invention also provide the additionalbenefit of reducing delay through a telephony system with multiplepacket generators, such as those depicted in FIG. 2A, FIG. 2B, FIG. 2Cand FIG. 5. For example, in a conventional telephony system utilizing asingle voice processing module, the time required to assemble a packetfor upstream transmission is the sum of the processing time for allchannels concatenated. In contrast, in embodiments of the presentinvention, the processing time advance is only for that of one channelbecause each of the voice processing modules carries out its compressionalgorithm in parallel.

[0089] Another benefit of the above-described system and method is thatthe host processor 502 need only sustain the packet rate of theconcatenated packets transferred from the voice processing modules 218,506 and 508. The concatenated packet rate is one-third the rate of MDUtelephony systems that do not use concatenation. Therefore, inembodiments of the present invention, it is possible for the hostprocessor to be sized as though the system were one third smaller thanconventional systems that do not use concatenation, resulting in a costsavings for a smaller, less capable processor system.

[0090] E. Conclusion

[0091] While various embodiments of the present invention have beendescribed above, it should be understood that they have been presentedby way of example only, and not limitation. For example, the presentinvention may be implemented in an MDU telephony system that supportsVoice Over Internet Protocol (VOIP) telephones instead of POTS phones.Furthermore, the present invention is not limited to the delivery ofpacket telephony services over a cable modem system but may beimplemented in any packet network system that has a reservation basedtransmission system. For example, the present invention may beimplemented in a fixed wireless communication system.

[0092] Accordingly, it will be understood by those skilled in the artthat various changes in form and details may be made to the embodimentsof the present invention that have been described herein withoutdeparting from the spirit and scope of the invention as defined in theappended claims. Thus, the breadth and scope of the present inventionshould not be limited by any of the above-described exemplaryembodiments, but should be defined only in accordance with the followingclaims and their equivalents.

What is claimed is:
 1. A telephony system, comprising: a first voiceprocessing module; a second voice processing module; a host processorcoupled to said first and second voice processing modules; and a buffercoupled to said first voice processing module, said second voiceprocessing module, and said host processor; wherein said first voiceprocessing module is adapted to receive first digital voice signals fromany one of a first plurality of subscriber lines, to compress said firstdigital voice signals to generate a first voice packet, and to transfersaid first voice packet to said buffer, wherein said second voiceprocessing module is adapted to receive second digital voice signalsfrom any one of a second plurality of subscriber lines, to compress saidsecond digital voice signals to generate a second voice packet, and totransfer said second voice packet to said buffer; and wherein said hostprocessor is adapted to assemble a packet comprising said first voicepacket and said second voice packet and to transmit said assembledpacket for delivery over a data network.
 2. The telephony system ofclaim 1, wherein said first voice processing module and said secondvoice processing module are further adapted to generate said first voicepacket and said second voice packet in parallel.
 3. The telephony systemof claim 1, wherein said host processor is adapted to transmit saidassembled packet for delivery over an HFC network.
 4. The telephonysystem of claim 3, wherein said host processor is adapted to transmitsaid assembled packet for delivery over an HFC network during anassigned upstream burst opportunity.
 5. The telephony system of claim 1,wherein said first voice processing module and said second voiceprocessing module each comprise a digital signal processor.
 6. Thetelephony system of claim 1, further comprising: a first audioprocessing module coupled to said first voice processing module; and asecond audio processing module coupled to said second voice processingmodule; wherein said first audio processing module is adapted to receivefirst analog voice signals from any one of said first plurality ofsubscriber lines and to convert said first analog voice signals intosaid first digital voice signals and wherein said second audioprocessing module is adapted to receive second analog voice signals fromany one of said second plurality of subscriber lines and to convert saidsecond analog voice signals into said second digital voice signals. 7.The telephony system of claim 6, further comprising: a first pluralityof subscriber line interface circuits coupled to said first audioprocessing module; and a second plurality of subscriber line interfacecircuits coupled to said second audio processing module; wherein eachone of said first plurality of subscriber line interface circuits isadapted to transmit analog voice signals from one of said firstplurality of subscriber lines; and wherein each one of said secondplurality of subscriber line interface circuits is adapted to transmitanalog voice signals from one of said second plurality of subscriberlines.
 8. The telephony system of claim 1, wherein said assembled packetcomprises physical layer overhead, media access layer overhead, saidfirst voice packet and said second voice packet.
 9. A telephony systemcomprising: a cable modem device, including a cable modem, a hostprocessor and a buffer; a first processing module coupled to said cablemodem device, wherein said first processing module includes a firstvoice processing module; and a second processing module coupled to saidcable modem device, wherein said second processing module includes asecond voice processing module; wherein said first voice processingmodule within said first processing module is adapted to receive firstdigital voice signals from any one of a first plurality of subscriberlines, to compress said first digital voice signals to generate a firstvoice packet, and to transfer said first voice packet to said buffer insaid cable modem device, wherein said second voice processing modulewithin said second processing module is adapted to receive seconddigital voice signals from any one of a second plurality of subscriberlines, to compress said second digital voice signals to generate asecond voice packet, and to transfer said second voice packet to saidbuffer in said cable modem device; and wherein said host processorwithin said cable modem device is adapted to assemble a packetcomprising said first voice packet and said second voice packet andwherein said cable modem within said cable modem device is adapted totransmit said assembled packet for delivery over a data network.
 10. Thetelephony system of claim 9, wherein said first voice processing modulewithin said first processing module and said second voice processingmodule within said second processing module are further adapted togenerate said first voice packet and said second voice packet inparallel.
 11. The telephony system of claim 9, wherein said first voiceprocessing module within said first processing module and said secondvoice processing module within said second processing module eachcomprise a digital signal processor.
 12. The telephony system of claim9, wherein said first processing module further comprises a first audioprocessing module coupled to said first voice processing module; whereinsaid second processing module further comprises a second audioprocessing module coupled to said second voice processing module;wherein said first audio processing module is adapted to receive firstanalog voice signals from any one of said first plurality of subscriberlines and to convert said first analog voice signals into said firstdigital voice signals; and wherein said second audio processing moduleis adapted to receive second analog voice signals from any one of saidsecond plurality of subscriber lines and to convert said second analogvoice signals into said second digital voice signals.
 13. The telephonysystem of claim 12, further comprising: a first plurality of subscriberline interface circuits coupled to said first audio processing module;and a second plurality of subscriber line interface circuits coupled tosaid second audio processing module; wherein each one of said firstplurality of subscriber line interface circuits is adapted to transmitanalog voice signals from one of said first plurality of subscriberlines; and wherein each one of said second plurality of subscriber lineinterface circuits is adapted to transmit analog voice signals from oneof said second plurality of subscriber lines.
 14. The telephony systemof claim 9, wherein said assembled packet comprises physical layeroverhead, media access layer overhead, said first voice packet and saidsecond voice packet.
 15. A method for reducing delay in a telephonysystem, comprising: receiving first digital voice signals from one of afirst plurality of subscriber lines; receiving second digital signalsfrom one of a second plurality of subscriber lines; compressing saidfirst digital voice signals using a first voice processing module togenerate a first voice packet; compressing said second digital voicesignals using a second voice processing module to generate a secondvoice packet; transferring said first voice packet to a buffer;transferring said second voice packet to a buffer; assembling a packetcomprising said first voice packet and said second voice packet; andtransmitting said assembled packet for delivery over a data network. 16.The method of claim 15, wherein said compressing said first digitalvoice signals and said compressing said second digital voice signals arecarried out in parallel.
 17. The method of claim 15, wherein saidtransmitting said assembled packet for delivery over a data networkcomprises transmitting said assembled packet for delivery over an HFCnetwork.
 18. The method of claim 15, wherein said transmitting saidassembled packet for delivery over an HFC network comprises transmittingsaid assembled packet for delivery over an HFC network during anassigned upstream burst opportunity.
 19. The method of claim 15, whereinsaid first voice processing module and said second voice processingmodule each comprise a digital signal processor.
 20. The method of claim15, further comprising: receiving first analog voice signals from one ofsaid first plurality of subscriber lines; receiving second analogsignals from one of said second plurality of subscriber lines:converting said first analog voice signals into said first digital voicesignals; and converting said second analog signals into said firstdigital voice signals.
 21. The method of claim 20, wherein saidreceiving said first analog voice signals from one of said firstplurality of subscriber lines comprises receiving said first analogvoice signals from one of said first plurality of subscriber lines via afirst subscriber line interface circuit; and wherein said receiving saidsecond analog voice signals from one of said second plurality ofsubscriber lines comprises receiving said second analog voice signalsfrom one of said second plurality of subscriber lines via a secondsubscriber line interface circuit.
 22. The method of claim 15, whereinsaid assembled packet comprises physical layer overhead, media accesslayer overhead, said first voice packet and said second voice packet.23. The method of claim 15, wherein said transferring said first voicepacket to a buffer comprises performing a first DMA transfer and whereinsaid transferring said second voice packet to a buffer comprisesperforming a second DMA transfer.
 24. The method of claim 15, whereinsaid first DMA transfer and said second DMA transfer are segmented DMAtransfers.